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Commit 59f9c7ce authored by Grzegorz Sluja's avatar Grzegorz Sluja Committed by Yalu Zhang
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Fix sequence number used by asterisk for outgoing RTP packets

There was no audio for 3-way conference when sRTP is used.
For 2-way calls frame->seqno is taken from DSP and is used by asterisk for the sequence number
in RTP headers. However for 3-way conference the sequence number is generated by asterisk and
it has to be greater than the previous value, otherwise libsrtp refuses to forward 'too old'
RTP packets.
parent 627a1902
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1 merge request!135Fix sequence number used by asterisk for outgoing RTP packets
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